asterisk users - Dec 2007

Monday December 31 2007
11:15PM 3 One Way Delay in Audio Over Analog
9:13PM 2 Problem with Polycom Soundpoint IP 320 Hardphone
5:56PM 0 How to use AddQueueMember with IAX2 peers?
5:36PM 3 Polycom Digit Map
2:42PM 0 Require IP Phones in Pakistan
10:31AM 1 PRI Crapping Out Regularly
6:05AM 1 app_echo.c
Sunday December 30 2007
3:48PM 1 Zap channels for HFC-S PCI card not responding
6:41AM 2 asterisk callerid
12:16AM 1 Looking for PSTN provider with unlimited inbound/outbound plan
Saturday December 29 2007
10:50PM 0 Cubix features at Asterisk: who is online
6:32PM 2 Cirpack KeepAlive packets causing SIP errors
3:27PM 1 Realtime & sip.conf
1:58PM 1 OT: Is Cisco 7960 SIP firmware same as 7940 SIP firmware?
1:20PM 0 Cisco IP phone 7975G + SCCP + Asterisk-1.4
9:19AM 5 Directories Used by Asterisk
8:14AM 0 IAX2 failed to authenticate; it uses wrong name
7:23AM 5 Digium Asterisk Appliance voicemail & logs
1:28AM 8 Asterisk 1.4 Fax
12:56AM 1 Not Able To tar zxvf zaptel-*.tar.gz
12:01AM 0 Voicemail App
Friday December 28 2007
11:26PM 0 Building prototype devices?
11:16PM 0 New voicemail vs. minivm
10:10PM 1 sip.conf & realtime
9:20PM 1 IVR help, please
6:23PM 0 call queuing not detecting caller hang up when call originates from voip provider
6:12PM 2 Problems with zaptel and HFC-S PCI card
4:36PM 1 Definity G3R and MWI
Thursday December 27 2007
10:38PM 8 New voicemail app (supports many interfaces, including Audix)
9:13PM 0 VoIP 2008 : wish list and predictions
4:33PM 1 How does Asterisk scale to 500-1000 phones?
4:10PM 2 No SMDI interfaces are available
3:07PM 3 CDR
2:13PM 1 Samsung iDCS 500R2 <PRI> Asterisk 1.4.*
10:01AM 0 zap transfer
9:37AM 1 application not load
9:32AM 3 Grandtream Conference issue
9:28AM 3 Performance Issues Degradation After 6 Calls
3:02AM 1 SIP Channel jitter buffer issue
Wednesday December 26 2007
10:11PM 0 Getting MOH after Attended Transfer
10:07PM 0 Fwd: Gotoif Time
6:53PM 1 smsq, Zaptel in UK
6:06PM 2 No cdr_csv after upgrade from 1.2.x to 1.4.x
5:36PM 2 Gotoiftime help
5:07PM 2 Unified Messaging On Thin Client / Terminal Server
5:03PM 1 ISDN BRI support with HFC-PCI cards?
2:32PM 2 Two lines for outgoing calls
10:53AM 0 autoservice.c
Tuesday December 25 2007
11:56PM 1 Softphone to be installed on the Mobile
6:50PM 1 cdr_adaptive_odbc and custom rdms fields
Monday December 24 2007
10:20PM 1 Marry Christmas and Happy New Year!!!
2:29PM 2 SIP Conference phones
2:00PM 1 sip.conf for
Sunday December 23 2007
10:53PM 1 Active Calls
6:05PM 3 OpenVox A800P01 and ZT_CHANCONFIG failed
12:31PM 0 Need some one to make a test call
10:51AM 1 Asterisk 1.2.26 badly broken?
1:34AM 2 PXE-bootable diskless Asterix distro?
Saturday December 22 2007
11:55AM 1 Sounds transscript / speech synthesis
7:51AM 7 Summary: Upgrading to Asterisk 1.4
5:40AM 5 call-limit in database
3:32AM 1 On-the-phone
1:49AM 0 Dead Incoming call - Sangoma A200
Friday December 21 2007
10:51PM 2 ODBC Voicemail and performance....
10:24PM 0 Control playback
8:06PM 0 SIP hangup on call proceeding message
7:28PM 2 best way for night ringer??
5:55PM 3 Polycom 330 beep on new VM
3:16PM 3 7970 CTLFile.tlv?
2:57PM 1 Send SIP 100 Trying instead of 183 Session Progress
2:55PM 1 Asterisk SIP handling - why 491 Request Pending response
2:38PM 0 Incoming CID change
12:13PM 2 Snom 370 buton Recordings
11:30AM 0 Resposta automàtica (was: asterisk-users Digest, Vol 41, Issue 67)
10:41AM 1 txfax not working with spandsp
Thursday December 20 2007
11:13PM 0 Failed Call Debugging?
9:38PM 0 Asterisk and Chan_h323: all calls are not going
9:29PM 2 Telco MWI Detection on TDM400 Interface?
7:53PM 0 H323 and Gatekeeper
7:43PM 1 MeetMeConference
7:01PM 3 Realtime: Should I say or should I go (now) ?
6:02PM 0 OT: VoIP SLA for SIP trunking - SMEs
3:39PM 1 put fxo channel before E1 channel?
3:32PM 2 Cisco 7961 new firmware stops reading configuration files
2:04PM 0 [VOIP-Users-Conference] Re: Digium: as of this a.m., one million Asterisk downloads this year
5:11AM 1 Asterisk.NET API --help required
4:33AM 7 ip phone suggestion for Asia?
1:37AM 1 Asterisk 1.4.15, Solaris and record command
1:12AM 2 turn off auto-seek extention - force use timeout
Wednesday December 19 2007
10:47PM 1 noun-verb vs verb-noun aka dogs black vs black dogs
10:12PM 3 Realtime logic in Asterisk
9:35PM 2 Bulk Reverse Phone Lookup
9:00PM 1 Asterisk
8:33PM 0 problems chanskype on ubuntu gutsy
7:20PM 5 Using * in extension name
4:03PM 0 Asterisk Realtime SIP rtcachefriends
2:53PM 1 IAX for asterisk to asterisk
2:39PM 2 asterisk with alsa
2:01PM 1 Asterisk and Codecs: g729 and g723
1:54PM 2 Download Asterisk
12:11AM 1 G.278 RTP conversation capture, please.
Tuesday December 18 2007
9:52PM 2 Asterisk/iaxclient IAX2 source port
9:27PM 2 resync linksys SPA9XX config file from Asterisk
9:22PM 1 Leading 0 in PRI outbound
8:04PM 0 Cisco 7970 BLF/Presence
8:03PM 0 AST-2007-027 - Database matching order permits host-based authentication to be ignored
7:50PM 2 Asterisk 1.4.16 and 1.2.26 released
7:35PM 1 SIP Anonymous auth
7:28PM 2 How to change sendmail return path
7:25PM 0 Softphone
7:21PM 0 Doorbell Siedle DCA 612 and Asterisk?
6:27PM 1 Call Recording on Hanup
4:43PM 1 Dropped Calls
3:16PM 4 AsteriskNOW release date???
2:20PM 1 How to automaticaly close calls when Asterisk didn't receive the bye request ?
12:37PM 1 Asterisk GUI - Call Waiting
10:47AM 4 All trunk are busy please try your call again later
9:59AM 4 Using MysqlPool Application 1.4
2:37AM 2 BLF trouble
1:50AM 0 Echo - when pressing digits
Monday December 17 2007
10:50PM 0 Automatic tests (was Re: Upgrade to Asterisk 1.4 - it's one year's old!)
10:28PM 2 Music On Hold
6:35PM 8 Queue calls drop to voicemail intermittantly
5:49PM 2 SIP call interrupted after 64 seconds
4:23PM 0 Friday @12 PM EST VOIP Users Conference + Aus/NZ/India/Japan conference event
1:07PM 1 Mail Test
11:57AM 0 Problem in Recording file on Hangup ?????
11:43AM 0 Cannot allocate memory
4:45AM 3 VoIP service providers/PSTN termination points
2:21AM 1 dial, answered and then hangup
1:53AM 3 Trixbox Phones Home
Sunday December 16 2007
8:09AM 1 Calling Party Category Field
7:41AM 0 LDAPget question, usage
4:42AM 0 Trixbox Arbitrary Command Execution Vulnerability
12:55AM 1 Newbie question: how to proxy the *real* caller-id on find-me/follow-me
12:25AM 1 Reputable company for SIP/IAX2 trunking
Saturday December 15 2007
5:57PM 2 DNS broken for ??
12:09PM 0 Open ITU G.107 Implementation to measure voice quality
12:01PM 0 OpenVox B800P and asterisk 1.4/ mISDN-1_1_7
10:57AM 17 Upgrade to Asterisk 1.4 - it's one year's old!
Friday December 14 2007
8:24PM 1 Asterisk to make multiple extensions simultaneous calls on a single telephone line
7:43PM 2 SIP fails to register
6:32PM 3 GUI for Asterisk: Call Flow
5:26PM 2 chan_h323 compilation
3:26PM 0 Monitor a queue
1:31PM 1 Asterisk Qeueu with static agent
12:49PM 2 Stange pause between extensions commands.
11:55AM 0 asterisk-users Digest, Vol 41, Issue 46
10:20AM 1 ZRTP + asterisk and Best Security Practice
8:06AM 0 G729 on PS3 Cell
6:41AM 2 Poor gsm playback
1:55AM 6 [Zaptel] Why no port to Windos?
12:30AM 0 Problem with TE205P with TeleWest in the UK
Thursday December 13 2007
11:16PM 0 Zaptel and released
11:12PM 0 Libpri 1.2.7 and 1.4.3 released
9:38PM 2 asterisk-users Digest, Vol 41, Issue 44
8:25PM 1 Asterisk 1.2.18 and Polycom phones not forwarding anymore
4:32PM 1 Cell Phone SMS
4:06PM 0 CallManager sip trunk - callerid name?
4:03PM 1 DID in Cape Town South Africa
12:49PM 2 How do I do this?
7:05AM 1 chan_mobile problems
6:52AM 0 Fwd: Re: Re: calls are getting dis
6:03AM 1 calls are getting disconnected automatically
1:52AM 0 Didnt get a frame from Channel and call gets
1:24AM 1 Sipura provisioning
Wednesday December 12 2007
11:42PM 1 Farward calls between 2 sip servers
10:10PM 1 Sip Version
8:28PM 4 TDM400 hangup issue in China
6:38PM 0 Can Local channels inhibit an Answer() until it is satisfied with the endpoint?
6:22PM 2 Linksys SPA962 with SPA932 unexpected reboots
5:32PM 1 Account codes in CDR
4:45PM 3 Polycom Paging
4:14PM 0 [Fwd: Request for testing SIP TCP/TLS]
2:14PM 1 Caller ID Issue
1:01PM 4 Enable/Disable Sip without registration
12:30PM 1 Asterisk B2BUA and Site to Site transfers
12:09PM 4 Call Center Setup on asterisk
8:32AM 0 asterisk-users Digest, Vol 41, Issue 38
8:08AM 3 Load Balancing over 2 E1 Lines
2:00AM 5 Call Quality Issues With 2 Trixbox's - Router Issue?
Tuesday December 11 2007
10:14PM 6 Most Stable version of Asterisk
10:07PM 1 Bribane bases contractor....
9:30PM 2 Aastra 480i CT
8:00PM 2 Check if SIP user is available or not ?
7:06PM 3 Any phone capable of displaying real time queue statistics?
6:47PM 0 VPN Client with the IP Phone and what its VPNServer
6:44PM 1 Asterisk not sending 200 OK
6:02PM 0 VPN Client with the IP Phone and what its VPN Server
5:51PM 1 Asterisk on IBM Netvista 2800 8364-EXX?
5:23PM 2 OT - Fax and anti-spam
5:06PM 0 asterisk-users Digest, Vol 41, Issue 35
4:37PM 0 new Asterisk installation with openvox 1.2 or 1.4?
3:26PM 1 merge gsm files
3:22PM 1 RFC3389 message
2:30PM 2 Recorded calls skipping
1:03PM 2 Unicall protocol error. Cause 32776
12:22PM 2 Iax and ZAP
12:03PM 1 rollback procedure requirements before asterisk upgrade
11:49AM 5 hi
10:09AM 4 X100P Fxo card headaches
8:16AM 2 VPN Client with the IP Phone, and what its VPN Server
8:14AM 1 Asterisk and NAT
8:07AM 1 Video Conference Or Server
7:39AM 0 Monitoring Asterisk 1.4.15
3:59AM 1 Appending two voice files
3:30AM 1 Fw: asterisk performance
3:08AM 0 line cut
1:58AM 5 SMS gateway recommendation
Monday December 10 2007
11:26PM 1 Didnt get a frame from Channel and call gets disconnected
10:04PM 1 Pickup re-invite
8:04PM 0 Checking Dial Status
7:08PM 0 Cisco 7911g Poor Audio Quality w/ Asterisk Voicemail and MOH
6:17PM 2 asterisk 1.4 with around 230 SIP connections
6:00PM 2 foneBRIDGE2 vs. foneBRIDGE2-EC
5:39PM 2 Dynamically change sip.conf properties.
5:30PM 0 text to speech
5:24PM 0 diferents events between ast1.2 & ast1.4 ??
3:06PM 2 SIP 7960 soft key customization?
1:16PM 2 asterisk linkedin group
1:01PM 0 Gateway doesn't ring
12:29PM 3 Graceful Asterisk Shutdown
12:24PM 2 Using Asterisk to connect 2 locations with legacy PBX
11:46AM 0 CAPI didn't get a frame | avoiding initial deadlock | multiple instances of Asterisk
11:15AM 0 Catalyst 2950 series with Asterisk + FAX
11:10AM 0 Asterisk + Cisco Call Manager Express
4:59AM 1 T.38 fax solution, opinions?
12:55AM 3 One server, multiple companies
Sunday December 9 2007
10:14PM 0 Dual-home Wifi/GSM phones for North America
9:18PM 1 [DB] Using SQLite instead of AST?
8:33PM 2 Don't enter a queue if no one is logged in
3:53PM 1 Installing/configuring TE120P debian way
Saturday December 8 2007
3:50PM 0 Asterisk CDR Variable
11:50AM 1 Direct Inward Dialing How To
10:34AM 0 Can't listen to voicemail message
9:30AM 5 using modem with asterisk
Friday December 7 2007
9:51PM 3 Using XML for configuration management, single-source-of-truth, etc.
8:53PM 1 Problem with the ring timeout in dial command for local extensions
8:04PM 1 Function vs. Application?
6:04PM 1 Limit participants in Meetme...
5:02PM 1 Show calls in progress
4:58PM 2 Polycom 601 stops ringing
4:50PM 0 AMQP Support for Asterisk?
4:28PM 2 Open Asterisk Exchange Project
4:21PM 2 Sidetone with Snom 370
2:40PM 0 Perspective on Asterisk
10:37AM 0 dtmf detection not working on sip trunks using asterisk-1.4.15
9:55AM 4 Any idea how making Asterisk "transparent"?
9:49AM 1 Pickup cmd
7:08AM 2 PHP AGI script
6:24AM 0 Asterisk is not adding Via field
5:14AM 2 7960 Won't Register Yet Multiple Attempts?
3:57AM 2 Where does the call go in the dialplan after a call disconnects
12:36AM 0 PRI: calling an Unallocated Number
12:09AM 2 scrubbing a call list to remove cell phone numbers
Thursday December 6 2007
11:40PM 2 Print CALLERID in CLI during "pri debug "
10:26PM 0 Perl FastAGI service port.
10:05PM 1 "Happy Birthday Asterisk"
9:13PM 1 Voicemail Question
8:35PM 0 VOIP Users Conference for Friday Dec 7 @ 12 Noon EST
8:04PM 3 CDR Function in Hangup Channel
7:22PM 4 Probems receiving 200ok message
6:20PM 3 Setting Multiple Values via func_odbc ...?
6:18PM 1 Dial() Macro option error in 1.4.15
3:37PM 3 Setting custom field in CDR
3:18PM 2 Cisco power injector with GXP2000 phones
3:00PM 2 Logging in and off sessions in the dialplan
2:17PM 0 Increasing the voice volume for digium card
2:11PM 2 Call Center Scenario -- take 2
1:19PM 3 Play Beep instead of MOH
1:07PM 0 IVR problem (Trixbox)
11:30AM 1 DeadAgi
10:26AM 0 Polycom call drops
9:53AM 2 GSM and CDMA Gateways
9:47AM 3 asterisk performance
9:44AM 0 C&W ISDN110
6:33AM 0 Can Asterix seperate the signalling and the media ip's with Quintum
3:47AM 1 Running AGI script if condition met?
2:51AM 0 TDm804B
2:33AM 1 s, CDR and NoCDR in v1.4.10.1
1:19AM 1 Polycom Soundpoint (NO LINE)
12:49AM 2 astunicall- packages and Sangoma A104D - ERROR
12:34AM 0 astunicall- packages and Sangoma A104D
Wednesday December 5 2007
10:42PM 1 Cisco 7960 to 2 SIP servers?
10:05PM 8 Strange ISDN-problem with incoming calls out of the same city
8:48PM 4 Asterisk server and DSCP QOS
7:25PM 3 No timezone in Voicemail email?
7:16PM 0 Asterisk and TDM400P
4:45PM 7 Asterisk SIP Microsoft Outlook Integration
3:47PM 3 Adtran supervision problems
3:45PM 0 redirected call failure
3:35PM 0 New feature: calling all bug marshals
3:18PM 1 SIP-Realtime and sip reload
2:57PM 0 Bad behaviour between X-Lite 3.0 and Asterisk
12:38PM 2 New feature: calling all bug marshals
9:25AM 2 Increasing the voice volume from the digium card
9:13AM 2 Text-To-Speech synthesizer--help required
8:49AM 1 [HELP] Problems with VOIP organization
8:40AM 1 Disturbance "noise" in the background for digium card
7:23AM 2 Multiple contacts.
6:03AM 5 New feature: calling all bug marshals
5:09AM 1 [Fwd: load test zap channels (in and out)]
5:03AM 0 Melbourne Asterisk meetup - again
Tuesday December 4 2007
9:55PM 0 Which rackable server for TDM2400 ?
8:17PM 1 Fax on asterisk
7:34PM 1 Call center scenario
6:17PM 4 Echo cancellation and DTMF from the Asterisk console?
4:24PM 1 New User
4:04PM 1 Soundcard necessary on an asterisk server toget output of playback()??
3:14PM 1 Voicemail box
2:41PM 0 System Specifications for 60 Extensions
2:39PM 0 Digium Support Comparison with Sangoma
2:02PM 2 pstn call waiting and zap
11:39AM 4 enable eyeBeam to accept only one call
9:43AM 1 Gtalk callerID
9:29AM 1 Explain AGI and AMI
7:21AM 1 Problem forwarding voicemail messages
4:49AM 0 Queue App - (1.4.15) free agents with callers waiting
4:35AM 0 Queue App - crash (1.4.15)
4:13AM 3 Phone with public address functionality
2:18AM 1 IBM x3400 w/ Digium TE220
Monday December 3 2007
10:10PM 0 Asterisk and Ekiga Chat
8:16PM 0 Adhearsion Install Fails.
6:20PM 2 Hoteling
6:14PM 1 MWI error
6:01PM 4 Soundcard necessary on an asterisk server to get output of playback()??
5:25PM 3 Anyone here using douBRI 2.0 ISDN ?
5:04PM 0 Click2call from Thunderbird
3:14PM 2 Red Alarm TE420 with E1s - R2
2:38PM 3 Replacing Skype with Asterisk Peering Servers - and Security
10:16AM 2 Problem: Using timelimit (L) and Macro (M) in Dial from AGI
9:34AM 1 SPA-3102 Registration Failed .. need advise
9:19AM 3 Underground Asterisk Command Set?
9:16AM 0 get REMOTE SIP extension status without calling it
8:11AM 2 MeetMe Conference on Asterisk-1.4.13
7:03AM 1 New VICIDIAL astGUIclient Release: 2.0.4
6:48AM 1 Oracle and asterisk
3:41AM 0 Asterisk 1.4.15 sip.conf register
2:02AM 1 Subject: Newb Question
Sunday December 2 2007
10:52PM 1 setting up two asterisk server as ss7 back to back.
9:42PM 2 Softswitch digim
8:18PM 2 Asterisk install beta testing/config help
7:48PM 0 When calling in via AGI, gsm sound file plays but sometimes drops out
7:22PM 1 T1 Timing Troubleshooting
6:22PM 1 What is the status and future of chan_mobile
5:38PM 0 Dictaphone Freedom interface to Asterisk ABE
4:49PM 2 Answering Machine Detection
2:30PM 1 Asterisk on Solaris
1:51PM 4 get SIP extension status without calling it
11:05AM 1 DID provider
5:35AM 1 Answer Machine/Fax/modem detection
12:09AM 2 Requiring a login to a phone
Saturday December 1 2007
5:50PM 2 Increasing the voice volume from the diguim cards
3:43PM 2 cdr_pgsql error in 1.4.15
1:35PM 0 Asterisk 1.4.15 Voicemail
1:22PM 1 Consulting/Integration Services Non-US & US *u
5:42AM 1 Asterisk & Cisco calling Name
5:13AM 1 REFER mesage extraction using SIP_HEADER
2:40AM 1 Registration state: Failed